This invention relates to wireless communications, and more particularly, to a method and apparatus for improving the quality of speech signals transmitted over a wireless communications system.
In order to maximize use of the limited available bandwidth for wireless communication services, various compression techniques and multiplexing techniques are used on the link between a base station, which is connected to a wireline telecommunications network, and a plurality of different users at mobile units each of whom are simultaneously communicating, via that base station, with other users who may be connected to either a wired or wireless network. Time Division Multiple Access (TDMA) and Code Division Multiple Access (CDMA) are examples of two well known multiplexing schemes used in wireless cellular and PCS systems. In TDMA, a time frame is subdivided into a plurality of time slots and a user""s mobile unit communicates in burst transmissions over a specific time slot at uplink and downlink carrier frequencies associated with the base station, which time slot assignment is made when a call commences. A receiver at a base station time-division multiplexes the plural digital speech signals directed to the plural mobile units into a downlink signal which is transmitted at a downlink carrier frequency to the mobile units. The receiver in each mobile unit then recovers the digital speech signal directed to itself in accordance with the time slot assigned to that mobile unit. Similarly, the uplink signal transmitted by each mobile unit is transmitted in an assigned time-slot, which time-slot assignment is used by the base station to appropriately recover the digital speech signal transmitted by the mobile unit. In CDMA, rather than assigning a mobile unit a specific time slot, a mobile unit is assigned a specific encryption code which is used to spread the spectrum of the coded speech signal over the channel. The plural mobile units that are simultaneously communicating with the base station are each assigned a different encryption/decryption code. The codes associated with each mobile unit are then used by the base station to multiplex the plural digital input speech signals into a downlink signal which is transmitted to all the mobile units. The mobile unit set then uses its associated decryption code to recover the particular digital speech signal directed to it from the multiplexed downlink signal transmitted by the base station. Uplink communication from the plural mobile units functions in a parallel manner.
In order to reduce the number of bits representing the coded digital speech signal, and therefore improve the efficiency of use of the available frequency bandwidth, speech compression techniques are used in wireless communication. The analog speech signal, which is normally sampled at, for example, an 8 kHz rate to produce a PCM bit stream, is subdivided into frames and compressed using an appropriate coding algorithm. In TDMA systems, a fixed rate vocoder, such as an ACELP (Algebraic Code Excited Linear Predictive) or VCELP (Vector-Sum CELP) coder, is used to compress the PCM samples. In CDMA systems, a variable rate CELP algorithm is used. Specifically, for CDMA, a speech encoder produces a variable rate output based on the speech activity of the input speech signal. During active speech periods, the speech encoder produces full rate 20 ms frames. During the silent periods, the speech encoder produces xe2x85x9 rate frames. During transition periods between the talking periods and silent periods, the speech encoder produces xc2xd rate and xc2xc rate frames. During these sub-rate frames, power consumption is reduced to a lower level than during full rate frames advantageously reducing the overall power output.
In the CDMA transmitter at either the mobile unit or at the base station, the PCM speech samples are broken down into 20 ms frames. The speech encoder uses an analysis-by-synthesis method to optimally determine the parameters for a given PCM speech frame input. For every 20 ms PCM speech frame input, the speech encoder produces a set of output parameters that represent the encoded frame. The speech encoder determines that set as a set of input parameters for an internal-to-the-encoder decoder, which minimizes the perceptual difference between the synthesized speech, which is output from the internal decoder, and the original input speech. An encoded full rate frame includes the following parameters: the linear predicative coding (LPC) parameters; pitch lag (L) and pitch gain (b) parameters; and codebook gain (G) and codebook index (I) parameters. At the receiving end, at the mobile unit or at the base station, the speech decoder receives each incoming encoded frame and converts each frame back into a sequence of PCM speech samples using the encoded frame. (see, e.g., TIA [Telecommunications Industry Association] IS-733-High-Rate [13 k] bits per second Speech Service Option).
As a consequence of the susceptibility of wireless links to interference and other inherent atmospheric conditions, a transmitted frame may not reach the receiver at either the base station or mobile unit set or may be severely corrupted by noise or interference. When a frame is xe2x80x9clostxe2x80x9d or is so corrupted with noise or interference as to be undecodable, it is marked by the receiver as being erased and no encoded parameters are supplied to the speech decoder. In order to minimize the perceptual effect of such an erased frame, a Frame Masking Algorithm is used to estimate the PCM samples for the erased frame using an extrapolation of the data from a previous frame. Thus, the speech decoder uses the previous values of the aforedescribed frame parameters to determine the current values of the erased frame. More specifically, for CDMA 13 k systems, the current values of the linear predictive coding (LPC) parameters are determined by decaying the LPC parameters of the previous frame, where the decaying coefficient is a function of the number of consecutive erasure frames. The current value of the pitch gain lag (L) is repeated from the previous frame; the current value of the pitch gain (b) is determined from the pitch gain of the last frame; the current value of the codebook gain (G) is determined by subtracting an appropriate integer from the previous value of G; and the codebook index (I) is determined randomly. A problem arises, however, when the previous frame to an erased frame is less than a full rate frame. When the previous frame is, for example, a xe2x85x9 rate frame, the resultant perceptive speech quality obtained by extrapolating parameters from the xe2x85x9 rate frame is poor.
In the prior art, the power of the signals transmitted on the downlink from the base station to the mobile unit set and on the uplink from the mobile unit to the base station is controlled to minimize power while maintaining an acceptable frame error rate. In particular, on the downlink, a power control algorithm makes a decision to increase or decrease the base station transmit power based on information provided by the mobile unit on the uplink. The mobile unit monitors the downlink and compiles statistics about the downlink frame error rate. This information is then conveyed back to the base station on the uplink to enable the base station to control its transmit power to maintain the desired downlink error statistics. On the uplink, the mobile unit controls its transmit power in direct response to power control order messages sent by the base station to the mobile unit on the forward link. The uplink power control algorithm determines the required direction of uplink transmit upward or downward power change and the magnitude of the change based on the history of the received frames to date. The history of the frames received on the uplink includes good frames, corrupt frames, and frame rate information.
Disadvantageously, such prior art power control schemes increase the power transmitted by the base station and the mobile unit only after corrupted frames have been received by the mobile unit and base station, respectively.
In accordance with the present invention, it has been recognized that as a sequence of frames are transmitted from the base station to the mobile unit or from the mobile unit to the base station, certain frames are more xe2x80x9cimportantxe2x80x9d from the voice quality perspective than others and thus should be individually transmitted in a manner that will better insure their proper reception at the receiving end of the radio link. By detecting these xe2x80x9ccriticalxe2x80x9d frames and improving their robustness before they are transmitted, each such critical frame will thus be less likely to be lost or corrupted as it is transmitted over the wireless transmission channel. As an example, in a CDMA embodiment, if the parameters of the previous frames can be used by the decoder to recreate the PCM samples of the current frame without any noticeable distortion, then the current frame can be characterized as a non-critical frame. On the other hand, if the parameters of the previous frame(s) cannot be used to recreate the PCM samples of the current frame without noticeable distortion, the current frame is characterized as a critical frame. Thus, if a full rate frame follows a partial rate frame, such as a xe2x85x9 rate frame, then that current full rate frame is identified as a critical frame before it is transmitted and its robustness is improved over non-critical frames before such a critical frame is transmitted. In an embodiment of the present invention in a CDMA system, a current frame is determined to be critical or non-critical by comparing the values of the frame parameters of the current frame with the corresponding frame parameters in a previous frame and forming a weighted sum of the corresponding differences, which is then compared with a threshold value. In this embodiment, once a frame is identified as being a critical frame, its robustness is improved by increasing its transmitted power. Thus, that critical frame, when transmitted at a higher power level, is less likely to be determined to be erased by a receiver, and more likely to reach the receiving end uncorrupted by noise. Alternative techniques for improving the robustness of a particular frame include improved channel coding that incorporate new methods of bit interleaving.